Setting up an MCU to send and receive calls with CUCM 6

Hierarchical Navigation

How do I set up a Cisco TelePresence MCU to send and receive SIP calls with Cisco Unified Communications Manager 6?

Follow these steps to configure the Cisco TelePresence MCU version 2.4 and later and Cisco Unified Communications Manager (CUCM) 6 in order to make outgoing calls from the Cisco TelePresence MCU and for it to be able to connect calls from CUCM 6 directly to conferences.

Creating a profile on CUCM

Before configuring SIP settings on the CUCM, you need to create a SIP profile. The following example creates a profile using most of the default settings:

Log in to the CUCM web interface.

Go to Device > Device Settings > SIP Profile and click Add New.

Enter a name for this profile (for example Standard SIP Profile).

Click Save.

Adding a user account on the CUCM

Go to User management > End User and click Add New.

Enter the User ID e.g. 6000.

Enter a password and PIN for this account.

For Last name enter MCU.

Click Save.

Adding the MCU as a phone device

CUCM 6 treats every device as a phone line, including the MCU.

Go to Device > Phone and click Add New.

For Phone Type select Third party SIP device (Advanced) and then click Next.

For MAC address do not use the MCU's port A MAC address: instead use the extension preceded by as many zeros as required to bring the total length to 12 digits e.g. 000000006000.

In the Association information section click Line [1] - Add a new DN link.
Note: To register an MCU conference you can create another user ID. Assign this ID to a different line (i.e. repeat the instructions in the previous section and then those in this section but in step 11 click the "Line 2" link). Then use it as the conference ID on the MCU. CUCM allows you to assign up to eight Directory Numbers (DNs). Alternatively, you can create a SIP trunk on the CUCM to the MCU if you need more eight conference IDs as explained later in this article.

Adding a participant into a conference

To call a participant into a conference, follow these steps. In this example the participant's endpoint is registered to CUCM with a user ID
of 6003.

On the MCU go to Conferences, select the conference and click Add participant.

For Address enter 6003.

For Call protocol select SIP.

Select Use SIP registrar.

Click Call endpoint.

In addition, any registered endpoint can dial into the MCU's auto attendant by calling the User ID that the MCU is registered with - 6000 in this example.

After completing these steps:

You can make outgoing SIP calls from the Cisco TelePresence MCU to SIP clients registered with the same CUCM

Clients registered on the CUCM can dial in to the Cisco TelePresence MCU's auto attendant or directly to a conference if the additional user was created for the conference

Creating a SIP trunk on CUCM for the MCU

A SIP trunk tells CUCM to send any call beginning with a certain prefix to the MCU's IP address. In this case, the MCU doesn't need to register with CUCM, but it accepts any calls that CUCM. Therefore you can reach any conference on the MCU without registering each one individually, so long as its ID starts with the correct prefix. Follow the instructions in this section. (Without a SIP trunk, you need to set up a user for the MCU to register with in order to reach its auto attendant plus one other user for each conference that you want to register as described earlier.)

Note: After the MCU is added as a SIP trunk, CUCM no longer accepts registration from the MCU. Therefore you still need to keep MCU SIP configuration but select No registration.

Login to the MCU and go to Settings > SIP.

For SIP registrar settings select No registration.

For SIP registrar address enter the IP address of the CUCM.

For SIP registrar type select Standard SIP.

For Username enter the User ID (6000 in our example).

Enter the password you provided earlier for this User ID.

For Outgoing transport select either UDP or TCP.

Click Apply changes.

On the CUCM:

A SIP trunk on CUCM requires a Media Termination Point. Check whether there is one by going to Service > Media Resources > Media Termination Point. Then leave the search field blank and click Find to check whether a Media Termination Point is found.

If one is found go on to the next section. If there is no Media Termination Point, go to the CUCM Serviceability web page.

Select the Cisco IP Voice Media Streaming App check box and then from the Tools menu select Service Activation.

Creating a SIP security profile for the trunk

Enter a name for the security profile, for example SIPTrunkToMCU and click Save.

Adding a trunk using the SIP security profile just created

Go to Device > Trunk and click Add New.

For Trunk type select SIP trunk and click Next.

For Device Name enter a name. The device name is used only internally in CUCM so it can be anything you want.

For Description, enter a description that will remind you of the purpose of this trunk.

For Device pool select Default.

In the SIP information section, for SIP Trunk Security Profile select SIPTrunkToMCU which you created earlier.

For DTMF signalling method select RFC 2833.

For Destination address add in the IP address of the MCU and for the port enter 5060.

Modify any other settings as needed for your CUCM installation.

Click Save and then click Reset.

Creating a Route Pattern

Add route patterns in CUCM to proxy calls to the Cisco TelePresence MCU using the SIP trunk that you have just created. In this example we configure CUCM to route incoming calls with prefix 21xx to the MCU.

Go to Call Routing > Route/Hunt > Route Pattern and click Add New.

Enter the route pattern 21XX.

For Gateway/Route list select the trunk name you have just created.

Click Save.

Configuring the conference ID to answer the calls from CUCM

On the MCU go to Conferences, and select the conference that you set up earlier.

Click Configuration.

For Numeric ID type in number within the range of the Route Pattern. The example route pattern was 21xx so you could use any four digit number beginning with 21, for example 2101.

Click Update conference.

The MCU is now ready to take incoming calls to 2101 and will connect the calling endpoint directly to the conference it called.